Maybe this is a very simple problem, but thought you can help on this.
I have a cisco 827-4V. It has 4 FXS ports. I have a SIP Configuration on it. I connected a telephone directly to port 1. I can make calls from the phone connected to port 1 of the 827-4V to anywhere with out any problem. However, when someone else from the SIP network dials to the number asigned to the 827-4V port 1, it gets bussy tone and on the other side, the telephone never rings.
I have other equipments like ATA's 186 on the same network working properly so I think that the problem might be on the configuration of the 827-4V. I'm attaching the current configuration. Can someone please take a fast look and tell me if I'm missing something on the voice-port configuration section or in the diail-peer to make the telephone ring and the person dialing to it not to get bussy tone???
Thanks in advance for your help!!!
Current configuration : 3960 bytes
no service pad
service timestamps debug uptime
service timestamps log uptime
aaa session-id common
enable secret 5 $1$rbY1$byVYnag8nrnJaqF6s5z5x.
enable password 7 130A1C1F
clock timezone GMT -6
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
class-map match-all voice
match access-group 102
ip address xxx.xxx.xxx.xxx 255.255.255.0 secondary
ip address xxx.xxx.xxx.xxx 255.255.255.0
ip nat inside
service-policy output politica1
hold-queue 100 out
no ip address
no ip mroute-cache
no atm ilmi-keepalive
dsl operating-mode auto
h323-gateway voip interface
h323-gateway voip id gk-mty-2 ipaddr xxx.xxx.xxx.xxx 1719 priority 120
h323-gateway voip h323-id adsl
h323-gateway voip tech-prefix 1#
interface ATM0.1 point-to-point
ip address xxx.xxx.xxx.xxx 255.255.255.252
ip nat outside
protocol ip xxx.xxx.xxx.xxx
ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx
ip route xxx.xxx.xxx.xxx 255.255.255.0 xxx.xxx.xxx.xxx
ip http server
ip http authentication local
ip pim bidir-enable
access-list 1 permit xxx.xxx.xxx.xxx 0.0.0.255
access-list 102 permit udp any any range 16384 32767
On the g/w pictured here, I'd issue a "debug voip ccapi inout" and then issue a call again the the gateway. Make sure to increase your terminal buffer. If you receive lots of data, then you know the call is getting to the box and you can then read the results. Look for "peer tag xxx". That will determine what dial-peer its matching on. At that point, it may be as simple as configuring your voip dial peer to be more specific than ....T.
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